All, My research indicates ANI is not really supported with SIP Channels or passed between SIP servers, even with setting function CALLERID(ANI). So the only place this applies is on PRI interfaces, when sending calls out a ZAP PRI you can set the ..
Im struggling to get the MWI set up on a few Polycom phones. The setup is like this. Ive got a few phones in the context called [company2_phones] and Ive got a few mailboxes in the voicemail context [company2]. Therefore, for each entry in sip.conf ..
i forgot to ask, how can i get the user number from a caller he is in a conference, i dont find a variable to us this for the current channel.
Only the command meetme list shows the usernumber, but i cant use this output.
All, I have a issue with park and pickup feature. I have asterisk 1.4.35 branch, Here is the scenario for the park and pickup. I have changed parking feature with *72 for my asterisk in features.conf. When i have inbound call it comes to one extens..
@ all, what is the best way to to use features like MeetmeCount without leaving the conference. I use Meetme(,X) and MEETME_EXIT_CONTEXT=context, but the problem is that the caller leave the Conference 🙁 Is it possible to press a key, without this obstac..
I want to create channel bank in this case:
i use the DYNAMIC_FEATURES (features.conf) to start a macro during a call, to start the Monitor application. In this macro i have a Playback to announce the recording. But the Playback play the soundfile only to the caller or to the callee depend..
I want set call limit for IAX2 users at the time incoming and outgoing,
Please help me how i can set call limit as asterisk support for SIP u..
On freepbx (GUI), whatever reason number fails we always get all circuits are busy audio. Does anybody know how to get far end audio when we dial wrong number or when its busy or unallocated number or failed with some other reason. Thanks..
I use the action Originate，i want the called first ringing，the called
answer,callee ringing.it can achieve?